Digital telecommunications system, program product for, and method of managing such a system

ABSTRACT

A digital telecommunications system, a method of managing a communications network in such a system and a program product for managing audio transmission in a digital communications system. A softswitch manages communications between devices at network endpoints, e.g., session initiation protocol (SIP) devices, and detects when communications include a non-human, e.g., an audio system, at an endpoint. The softswitch selects conversational communications for calls between voice devices and messaging communications parameters with lower overhead for communications with an audio system, e.g., messaging systems such as voice mail.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention is related to reducing transmission overhead indigital telecommunications systems and networks and more particularly,to reducing human to machine overhead in Voice over Internet Protocol(VoIP) based telecommunications.

2. Background Description

State of the art telecommunication systems are digital and, frequently,use Internet Protocol (IP) based communications. Unlike analog voicechannels with a continuous analog signal, an IP communications systemsegments audio data, encodes and packetizes the segments and transmitsthe encoded IP packets between network entities in a connectionlesstransfer. Bearing in mind that the human ear has a range of no more than20 Hertz (20 Hz)-20 KHz and typical telecommunications channels may beonly on the order of hundreds of Hz, audio occupies a very small portionof a typical IP communication. Since the minimum sampling rate for asignal to avoid aliasing is twice the highest signal frequencycomponent, a 500 Hz frequency component produces 1000 samples (e.g. 1KBytes or, for 8 bit samples, 8000 bits) per second. If a single 1 KBytesample is sent every second, there is at least a one second (1 s)latency at the receiving end that is further extended by anytransmission delays. Delays between samples cause gaps in the receivedaudio, as well as adding to the latency. So, using packets that are toolarge and system delays that cause gaps in the transmission such (e.g.,causing packet spacing to not be uniform, causes the receiving end audioto halting, fragmented and/or choppy, i.e., what is commonly discussedwith Quality of Service (QoS) issues. Trans-Atlantic TV news reportsprovide common examples of this.

So, standards have been developed and promulgated for Voice over IP(VoIP) communications to insure that typical IP networks compensate fortransmission delays and address QoS issues. These standards selectadequately small size for audio segments for encoding as relativelysmall packets and select transmitting those encoded small packets at arelatively high frequency such that decoding and transmission delays areunnoticeable or, at least, tolerable.

G729 is one such standard audio data compression algorithm for VoIP,wherein raw audio is segmented into 10 millisecond segments and eachsegment is compressed in an IP packet. RFC 3551 defines a net audio datastream for a G729 code/decode (codec) with an 8-kbit/sec data rate. See,e.g., www.apps.ietf.org/rfc/rfc3551.html#sec4.2. Normally, VoIP devicesthat use the G729 codec, are configured to default for a payload of20-Bytes/packet with 50-Packets/sec to achieve this 8-kbit/sec datarate. Id.

Real-time Transport Protocol (RTP) packets, for example, include headersthat used by IP networks for identification and routing. So, regardlessof packet size, 20 or 1000 Bytes, each packet has a fixed overhead.Since packet headers are in addition to and not part of the audio andeach packet, regardless of size, includes the same size header, smallerpackets incur higher overhead than larger packets. Small packets andhigh transmission frequency require more channel bandwidth and packetrouting and desegmentation requires higher processing capability, i.e.,more Machine Instructions per Second (MIPS). Consequently, VoIPcommunications require a relatively high level of system resources.

Messaging systems, such as voice mail, are common features in moderntelecommunications systems. Typically, unanswered calls are routed tovoice mail where the caller is greeted with an announcement and/or avoice recorder facility. Although RFC 3551 allows relaxed transfercharacteristics that accept higher packetization delays fornon-interactive applications (machine-to-machine or browser-to-browser)such as streaming audio/video, IP radio, lectures (webinars) or forlinks with severe bandwidth constraints, those relaxed transfercharacteristics are set by the originating device, e.g., the source ofthe stream. Such streams based on any G7xx codec use very large RTPpackets and may have very large spooler buffer at the receiving end,that spools, perhaps, a few seconds of the media packets.

However, normal VoIP telephony communications between devices in stateof the art VoIP communications systems almost always originate with ahuman, e.g., someone calling from a VoIP phone. The VoIP phone selectstransfer parameters for a voice call, i.e., human-to-human. Thus, thesehuman originated calls consume the same level of resources regardless ofwhether a call is between humans or with a machine, e.g., voice mail.However, reducing the overall consumption of system resources, wouldallow one to use lower performance systems to handle the same capacity,or achieve increased system capacity for the same system.

Thus, there is a need for reducing VoIP communications overhead,optimizing packet size in VoIP communications system and for minimizingcall resource consumption in VoIP communications, especially for humanto machine VoIP communications.

SUMMARY OF THE INVENTION

It is a purpose of the invention to reduce VoIP communications overhead;

It is another purpose of the invention to minimize call resourceconsumption in VoIP communications;

It is yet another purpose of the invention to optimize VoIPcommunications packet size in human to machine VoIP communications;

It is yet another purpose of the invention to optimize VoIPcommunications packet in human to machine VoIP communications forreduced VoIP communications overhead and minimized call resourceconsumption.

The present invention relates to a digital telecommunications system, amethod of managing a communications network in such a system and aprogram product for managing audio transmission in a digitalcommunications system. A softswitch manages communications betweendevices at network endpoints, e.g., session initiation protocol (SIP)devices, and detects when communications include a non-human, e.g., anaudio system, at an endpoint. The softswitch selects conversationalcommunications for calls between voice devices and messagingcommunications parameters with lower overhead for communications with anaudio system, e.g., messaging systems such as voice mail.

BRIEF DESCRIPTION OF THE DRAWINGS

The foregoing and other objects, aspects and advantages will be betterunderstood from the following detailed description of a preferredembodiment of the invention with reference to the drawings, in which:

FIG. 1 shows an example of an Internet Protocol (IP) communicationssystem including a digital call capable network with a machine audioresponse capability system according to a preferred embodiment of thepresent invention;

FIG. 2 shows an example of message flow for a call from endpoint toendpoint, and human to machine communications, e.g., voice mail.

DESCRIPTION OF PREFERRED EMBODIMENTS

Turning now to the drawings and more particularly, FIG. 1 shows anexample of an Internet Protocol (IP) communications system 100 includinga digital call capable network 102, e.g., capable of Voice over IP(VoIP) communications, with an audio system 104, according to apreferred embodiment of the present invention. The system includes EndPoints (EP) 106, 108, 110 with connected digital telephony devices(e.g., VoIP phones) and Multimedia Terminal Adapters (MTA), e.g.,keysets or session initiation protocol (SIP) phones. Since a networkdevice defines an EP, each EP and a device(s) at the EP are referred toherein interchangeably. A suitable proxy server 112 provides a routerfunction to private network 102. A gateway 114, e.g., a state of the artmedia gateway, connects the network externally, e.g., to a publicswitched telephone network/public land mobile network (PSTN/PLMN). Apreferred softswitch 116 manages network communications.

A preferred softswitch 116, e.g., a Media Gateway Controller (MGC)located in a data center, manages calls to/from keysets 106, 108, 110from/to each other, to/from or through the gateway 114 and managescommunications with the audio system 104. The audio system 104 may be asystem with an audio response capability, e.g., for announcements and/orvoice mail (hereinafter voice mail system for simplicity of discussion).In particular, a preferred softswitch 116 detects whether networkcommunications are conversational for real-time human-to-humancommunications, such as calls with/between the digital telephony devices106, 108, 110; or that the network communications are messaging with oneside being a machine (e.g., voice mail system 104). Preferably, thevoice mail system 104 provides transfer characteristics (conversationalor messaging) for the connection according to a preferred embodiment ofthe present invention to reduce overall overhead in communications.Optionally, if the voice mail system 104 does not include a capabilityfor indicating messaging transfer characteristics, upon detecting thatthe voice mail system 104 is on one side of the communications and thatthe call is human-to-machine, the softswitch 116 selects appropriatemessaging transfer characteristics based on that detection.

Preferably, any network provided devices such as gateway 114,voice-mail-server 104 and announcement-machines (not shown) are capableof standard available G7xx codecs, e.g., GSM, G.723, G.729, G.711.Further, digital telephony devices 106, 108, 110 may be sophisticatedprocessor based VoIP devices that also have such standard codeccapability and configured for conversational communications inreal-time. However, digital telephony devices 106, 108, 110 may beconsumer or user devices with a capability of a subset of thoseavailable conversational codecs. Thus, that capability may restrict thenetwork by for codec selection to optimize bandwidth usage.

Since conversational communications are unnecessary with voice mailsystem 104 (i.e., the comfort of uninterrupted, unhalting (not choppy),undelayed human to human conversation is unnecessary), the preferredsoftswitch 116 signals messaging transfer characteristics forcommunications with voice mail system 104. Specifically, the preferredsoftswitch 116 selects messaging transfer characteristics to minimizeoverhead. In particular, according to a preferred embodiment of thepresent invention, the preferred softswitch 116 selects messagingtransfer characteristics with a larger packet size and/or lower transferfrequency for human to machine communications. Optionally, the preferredsoftswitch 116 also lowers packet priority for such human to machinecommunications.

Optionally, a preferred system 100 may compensate transmission jitter,by spooling the incoming packets into a buffer or local storage at thereceiver (e.g., 118 at 110) before decoding and playing audio. Whilethis spooling may itself cause additional delays in playback (e.g.,transmission and spooling delays), effectively delaying communication,the delay is acceptable, tolerable and may even be unnoticeable. Delayedaudio is most noticeable and intolerable for two communicating partnersthat also have direct visible contact each other. However, it has beenfound that, for two partners communicating at a distance and unable tosee each other, delays of up to 200 ms are tolerable. So, regardingtransmission and spooling delays, the packet transfer rate (number ofpackets per second), the packet size, and the receiver's spool buffersize are parameterized and configured for a minimal audio delay and,more particularly, not to exceed 200 ms audio delay.

In a typical state of the art IP network for example, with User DatagramProtocol (UDP) transmission selected, payload packets transfer inReal-time Transport Protocol (RTP). The RTP overhead includes IP routinginformation of 20-Bytes, a UDP identifier of 8-Bytes and an RTPdescription of 12-Bytes. So, RTP requires a transport header of 40 Bytesper packet (Bpp) to transport a 20-Byte G729 payload. Consequently,using the G729 codec for conversational communications between digitaltelephony devices 106, 108, 110 at system endpoints, the efficiency is20 Bpp (net)/60 Bpp (gross)=33%. By contrast where conversationalcommunications are unnecessary, the softswitch 116 may signal, forexample, a human to machine (h/m) messaging transfer at 500 Bytes perpacket and the transfer rate at 2 packets per second (a period of 500ms). The efficiency jumps to 93% for human to machine messaging, i.e.,500 Bpp (h/m net)/(500 Bpp+40 Bpp)=500/540=93%. The overall efficiencyincreases 60%, 93% vs. 33%. It should be noted that the above packetsizes and transfer rates, as well as conversational and messagingcodecs, transfer characteristics and protocols are indicated for exampleonly and not intended as a limitation.

FIG. 2, with reference to FIG. 1, shows an example of message flow 120for a call from endpoint “A” 110 to endpoint “B” 108, subsequentlyaccompanied by human to machine communications (e.g., voice mail)according to a preferred embodiment of the present invention. In normaltwo-way communications both sides of a communication exchange respectiveSession Description Protocol (SDP) parameters, e.g., in one or more SIPmessages. See, e.g., RFC4566, “SDP: Session Description Protocol,”www.apps.ietf.org/rfc/rfc4566.html. With any communication, theoriginating/transmitting device selects one or more codecs. Theselection is included in the SDPs (SDP-A and SDP-B in this example)originating from the device that indicates support by that device ofone/multiple codec(s). The receiving device or, the partner to thecommunications, indicates which supported codec it can encode/decode.

So in the real-time communications portion of this example, endpoint “A”110 initiates a call by sending 122 an INVITE message to softswitch 116.The INVITE message includes SDP-A information that describes streamingmedia initialization parameters for the voice-to-voice/human-to-humanconversational communications in real-time, e.g., indicating the G729codec. The softswitch 116 forwards 124 the INVITE message to the calledendpoint “B” 108. The called endpoint “B” 108 responds 126 to softswitch116 with a 200 OK message that also includes SDP-B information. TheSDP-B information also indicates G729 codec for a normal conversationalconnection. The softswitch 116 forwards 128 the 200 OK message tocalling endpoint “A” 110, at which point the endpoints 108, 110 enter anormal talk state 130. In the normal talk state 130, the endpoints 108,110 exchange packets 132, e.g., using the G729 codec in RTP at 20 Bpp.

After a period of conversation (talk) 130, however, the called user atendpoint “B” 108 decides to transfer the rest of the conversation tovoice mail (VM) in this example. For example, a secretary/receptionistanswers 126 and responds, “Mr. Bond isn't in. Would you like his voicemail?” When the transfer begins when the called endpoint “B” 108 sends arefer message 134 to softswitch 116, referring the caller to voice mail.Then, the softswitch 116 forwards 136 the INVITE message to voice mailsystem 104 and terminates 138 the connection to the called endpoint “B”108 with a 202 & BYE message. The INVITE message at 136 still includesthe SDP-A information for conversational communications. Responding tothe INVITE message, the voice mail system 104 sends 140 a 200 OK messageto the softswitch 116 that, in this example, includes SDP,VMinformation. The “,VM” is a signal that the nature of the call ischanging from conversational to messaging. Alternately, the softswitch116 may be aware that endpoint 104 is a voice mail system and,therefore, that the call is changing from conversational to messaging.In this alternate example, the 200 OK message 140 from the voice mailsystem 104, may omit the “,VM” signal.

The SDP,VM information includes an indication of messaging transfercharacteristics for a human to machine connection that does not requireconversational communications and QoS. The messaging transfercharacteristics specify the messaging parameters as selected for voicemail messaging, e.g., larger packet size and lower transfer rate forless bandwidth than conversational communications. The softswitch 116forwards 142 a re-INVITE message to the calling endpoint “A” 110. If theendpoint 104 omitted the “,VM” signal, the softswitch 116 inserts thesignal and, either way, that the softswitch 116 forwards 142 a re-INVITEmessage that includes the SDP,VM information indicating the messagingtransfer characteristics. The calling endpoint “A” 110 responds 144 witha 200 OK message to the softswitch 116 that includes SDP-A information,also indicating the messaging transfer characteristics for the human tomachine connection.

Upon receiving 144 the 200 OK message the softswitch 116 opens amodified talk state 146 between the calling endpoint “A” 110 and voicemail system 104. In the modified talk state 146, the calling endpoint“A” 110 and voice mail system 104 exchange messaging packets 148 at themore efficient, larger packet size and less frequent transfer rate,e.g., in RTP at 500 Bpp. So as noted hereinabove for this example,conversational, human-to-human communications maintain 33% efficiency;with the human to machine messaging communications selecting largerpackets and the higher h/m transfer rate applied to both ends 104, 110,the efficiency is 93% for voice mail communications.

Advantageously, the present invention provides more efficient resourceconsumption for communications between a human and a machine. Unlikecurrent VoIP systems with fixed media transfer parameters (e.g., packetsize and transfer rate) regardless of endpoints communicating, thepresent invention selects media packet sizes tailored for thecommunications needs and capabilities of the endpoints. Connections thatrequire higher QoS (human-to-human connections) have higher performancetransfer parameters; connections with less demanding requirements(human-to-machine connections) have more relaxed transfer parameters.This improvement is achieved by signaling with call processing signalsthat indicate to the softswitch whether a connection is a default(human-to-human) connection or a less demanding (human-to-machine (e.g.,for announcements, voice-mail and/or answering machines)). Alternately,the softswitch may add transfer parameters and information afterdetecting that a machine is at least one side of a connection. Thesoftswitch may detect based on received signals or administrative dataabout communicating partners. System endpoints reconfigure codec datatransmission to adapt to the particular situation. This benefits bothend points in any human to machine connection, i.e., the phone used bythe human and the voice-mail machine.

While the invention has been described in terms of preferredembodiments, those skilled in the art will recognize that the inventioncan be practiced with modification within the spirit and scope of theappended claims. It is intended that all such variations andmodifications fall within the scope of the appended claims. Examples anddrawings are, accordingly, to be regarded as illustrative rather thanrestrictive.

1. A digital telecommunications system comprising: one or more voicecommunications devices, each at one of a plurality of communicationsnetwork endpoints; an audio system at one of said plurality ofcommunications network endpoints; and a softswitch managingcommunications between devices at said plurality of communicationsnetwork endpoints, conversational communications parameters having afirst overhead being selected for calls with said one or more voicecommunications devices, and messaging communications parameters having asecond overhead lower than said first overhead being selected forcommunications with said audio system; and wherein said digitaltelecommunications system is a Voice over Internet Protocol (VoIP)system, said audio system is a messaging system, said voicecommunications devices are VoIP communications devices; and wherein afirst packet size and a first transfer rate is selected forconversational communications and a second packet size and a secondtransfer rate is selected for messaging communications with saidmessaging system, said second packet size being larger than the firstpacket size and said second transfer rate being lower than the firsttransfer rate.
 2. The digital telecommunications system as in claim 1,wherein communications between endpoints have an 8-kbit/sec data rate.3. The digital telecommunications system as in claim 1, wherein thefirst packet size is 20-Bytes/packet, the first transfer rate is50-Packets/sec, the second packet size is 500-Bytes/packet, and thesecond transfer rate is 2-Packets/sec.
 4. The digital telecommunicationssystem as in claim 1, wherein said messaging system is a voice mailsystem and at least one VoIP communications device includes storagespooling messaging communications from said voice mail system.
 5. Thedigital telecommunications system as in claim 1, wherein said softswitchselectively identifies whether communications at a voice communicationsdevice are with said messaging system and selects said messagingcommunications parameters responsive to identifying said messagingsystem being at one end of a connection.
 6. The digitaltelecommunications system as in claim 1, wherein said messaging systemand said VoIP communications devices are Session Initiation Protocol(SIP) devices, said softswitch determining from SIP messages from saidSIP devices whether said messaging system is responding to an INVITEmessage from a SIP phone and sending a SIP re-INVITE message withmessaging Session Description Protocol (SDP) parameters to said SIPphone whenever said messaging system is responding.
 7. A method ofmanaging a communications network, said method comprising the steps of:a) signaling a call from a first network endpoint of a plurality ofnetwork endpoints to a digital communications device connected toanother of said plurality of network endpoints; b) determining whether aresponse to said signaled call from a responding network endpoint isfrom an audio system; c) selecting communications parameters forcommunications between said first network endpoint and said respondingnetwork endpoint responsive to said determination; and d) opening a talkstate between said first network endpoint and said responding networkendpoint, communications between said first network endpoint and saidresponding network endpoint transferring across said network responsiveto said selected communications parameters; and wherein said audiosystem is a messaging system and the step (c) of selectingcommunications parameters comprises selecting conversationalcommunications parameters having a first overhead for calls when a voicecommunications device is at said responding network endpoint, andselecting messaging communications parameters having a second overheadlower than said first overhead when said messaging system is at saidresponding network endpoint; and wherein said conversationalcommunications parameters indicate a first packet size and a firsttransfer rate for conversational calls, said messaging communicationsparameters indicate a second packet size and a second transfer rate formessaging communications with said messaging system, said second packetsize being larger than the first packet size and said second transferrate being lower than the first transfer rate.
 8. The method of managinga communications network as in claim 7, wherein said communicationstransfer in step (d) comprises a User Datagram Protocol (UDP)transmission with payload packets transferring in Real-time TransportProtocol (RTP), conversational communications with voice communicationsdevices having a 33% efficiency, and messaging communications with saidmessaging system having a 93% efficiency.
 9. The method of managing acommunications network as in claim 7, wherein said communications systemis a Voice over Internet Protocol (VoIP) system and said communicationstransfer in step (d) is performed with an 8-kbit/sec data rate.
 10. Themethod of managing a communications network as in claim 9, wherein step(d) further comprises spooling transferred communications in storage ina VoIP communications device at said first network endpoint, saidspooled communications being played at said VoIP communications device.11. The method of managing a communications network as in claim 9,wherein signaling comprises sending Session Initiation Protocol (SIP)messages, a softswitch determining in step (b) responsive to SIPmessages from endpoints and in step (c) sending a SIP re-INVITE messagewith messaging Session Description Protocol (SDP) parameters to saidfirst network endpoint whenever said messaging system is at saidresponding network endpoint.
 12. The method of managing a communicationsnetwork as in claim 7, wherein the step (c) of selecting communicationsparameters selects a lower priority for communications with said audiosystem at said responding network endpoint than for communications witha voice communications device at said responding network endpoint.
 13. Acomputer program product for managing audio transmission in a digitalcommunications system, said computer program product comprising anon-transitory computer readable medium having computer readable programcode stored thereon, said computer readable program code comprising:computer readable program code for receiving Session Initiation Protocol(SIP) messages from network endpoints; computer readable program codefor forwarding SIP messages to said network endpoints; computer readableprogram code for determining whether a received SIP message is aresponse to a forwarded SIP message and whether said response originatesfrom an audio system at a network endpoint; computer readable programcode for selecting communications parameters for communicating endpointsresponsive to said determination, conversational communicationsparameters having a first overhead when a voice communications device isat said responding network endpoint, and selecting messagingcommunications parameters having a second overhead lower than said firstoverhead when said audio system is at said responding network endpoint;and computer readable program code for opening a talk state between saidcommunicating endpoints, communications between said communicatingendpoints transferring across said network responsive to said selectedcommunications parameters; and wherein said audio system is a messagingsystem, said conversational communications parameters indicate a firstpacket size and a first transfer rate for conversational communicationsand said messaging communications parameters indicate a second packetsize and a second transfer rate for messaging communications with saidmessaging system, said second packet size being larger than the firstpacket size and said second transfer rate being slower than the firsttransfer rate.
 14. The computer program product as in claim 13, whereinsaid computer readable program code for selecting communicationsparameters comprises computer readable program code for sending are-INVITE SIP message with messaging Session Description Protocol (SDP)parameters responsive to determining whether said response originatesfrom a messaging system.
 15. The computer program product as in claim14, wherein said computer readable program code for opening a talk statecomprises computer readable program code for User Datagram Protocol(UDP) transmission with Real-time Transport Protocol (RTP) payloadpackets, conversational communications between SIP phones having a 33%efficiency, and messaging communications with said messaging systemhaving a 93% efficiency.
 16. The computer program product as in claim13, wherein said computer readable program code for selectingcommunications parameters comprises computer readable program code forselecting a lower priority for communications with said audio system atsaid responding network endpoint than for communications with a voicecommunications device at said responding network endpoint.